1. Field of the Disclosure
The technology of the disclosure relates generally to Web Real-Time Communications (WebRTC) interactive sessions.
2. Technical Background
Web Real-Time Communications (WebRTC) represents an ongoing effort to develop industry standards for integrating real-time communications functionality into web clients, such as web browsers, to enable direct interaction with other web clients. This real-time communications functionality is accessible by web developers via standard markup tags, such as those provided by version 5, of the Hypertext Markup Language (HTML5), and client-side scripting Application Programming Interfaces (APIs), such as JavaScript APIs. More information regarding WebRTC may be found in “WebRTC: APIs and RTCWEB Protocols of the HTML5, Real-Time Web,” by Alan B. Johnston and Daniel C. Burnett (2012, Digital Codex LLC), which is incorporated herein in its entirety by reference.
WebRTC provides built-in capabilities for establishing real-time video, audio, and/or data streams in both point-to-point interactive sessions and multi-party interactive sessions. The WebRTC standards are currently under joint development by the World Wide Web Consortium (W3C) and the Internet Engineering Task Force (IETF). Information on the current state of WebRTC standards can be found at, e.g., http://www.w3c.org and http://www/ietf.org.
To establish a WebRTC interactive session (e.g., a real-time video, audio, and/or data exchange), two web clients may retrieve WebRTC-enabled web applications, such as HTML5/JavaScript web applications, from a WebRTC application server. Through the web applications, the two web clients engage in a media negotiation to communicate and reach an agreement on parameters that define characteristics of the WebRTC interactive session. This media negotiation is known as a WebRTC “offer/answer” exchange. Once the WebRTC offer/answer exchange is complete, the web clients may then establish a direct peer connection with one another, and may begin a real-time exchange of media or data. The peer connection between the web clients typically employs the Secure Real-time Transport Protocol (SRTP) to transport real-time media flows, and may utilize various other protocols for real-time data interchange.
In multi-party WebRTC interactive sessions, each participating web client may be directly connected to every other participating web client in what is referred to as a “full mesh” or “fully distributed” architecture. However, the “full mesh” architecture may impose significant burdens on computing resources and bandwidth utilization. Multi-party WebRTC interactive sessions may also be handled by a central media server to which every web client is connected. Implementing such a central media server may pose challenges due to new media extensions employed by WebRTC, as well as variations between implementations of a WebRTC media stack in different web client types and client versions.